/external/chromium_org/third_party/webrtc/modules/audio_coding/neteq/tools/ |
H A D | rtp_generator.cc | 20 WebRtcRTPHeader* rtp_header) { 21 assert(rtp_header); 22 if (!rtp_header) { 25 rtp_header->header.sequenceNumber = seq_number_++; 26 rtp_header->header.timestamp = timestamp_; 28 rtp_header->header.payloadType = payload_type; 29 rtp_header->header.markerBit = false; 30 rtp_header->header.ssrc = ssrc_; 31 rtp_header->header.numCSRCs = 0; 32 rtp_header 18 GetRtpHeader(uint8_t payload_type, size_t payload_length_samples, WebRtcRTPHeader* rtp_header) argument 47 GetRtpHeader(uint8_t payload_type, size_t payload_length_samples, WebRtcRTPHeader* rtp_header) argument [all...] |
H A D | neteq_performance_test.cc | 57 WebRtcRTPHeader rtp_header; local 63 rtp_gen.GetRtpHeader(kPayloadType, kInputBlockSizeSamples, &rtp_header); 80 lost = ((rtp_header.header.sequenceNumber - 1) % lossrate) == 0; 85 rtp_header, input_payload, payload_len, 94 &rtp_header);
|
H A D | rtp_generator.h | 39 // Writes the next RTP header to |rtp_header|, which will be of type 44 WebRtcRTPHeader* rtp_header); 73 WebRtcRTPHeader* rtp_header) OVERRIDE;
|
/external/chromium_org/third_party/webrtc/modules/rtp_rtcp/source/ |
H A D | rtp_receiver_video.cc | 50 int32_t RTPReceiverVideo::ParseRtpPacket(WebRtcRTPHeader* rtp_header, argument 60 rtp_header->header.sequenceNumber, 62 rtp_header->header.timestamp); 63 rtp_header->type.Video.codec = specific_payload.Video.videoCodecType; 66 payload_length - rtp_header->header.paddingLength; 69 return data_callback_->OnReceivedPayloadData(NULL, 0, rtp_header) == 0 ? 0 74 RtpDepacketizer::Create(rtp_header->type.Video.codec, data_callback_)); 80 rtp_header->type.Video.isFirstPacket = is_first_packet; 81 return depacketizer->Parse(rtp_header, payload, payload_data_length) ? 0 : -1; 110 int32_t RTPReceiverVideo::BuildRTPheader(const WebRtcRTPHeader* rtp_header, argument [all...] |
H A D | rtp_format_vp8_unittest.cc | 414 WebRtcRTPHeader rtp_header; local 415 memset(&rtp_header, 0, sizeof(rtp_header)); 418 EXPECT_TRUE(depacketizer_->Parse(&rtp_header, packet, sizeof(packet))); 420 EXPECT_EQ(kVideoFrameDelta, rtp_header.frameType); 421 VerifyBasicHeader(&rtp_header, 0, 1, 4); 423 &rtp_header, kNoPictureId, kNoTl0PicIdx, kNoTemporalIdx, kNoKeyIdx); 435 WebRtcRTPHeader rtp_header; local 436 memset(&rtp_header, 0, sizeof(rtp_header)); 468 WebRtcRTPHeader rtp_header; local 486 WebRtcRTPHeader rtp_header; local 505 WebRtcRTPHeader rtp_header; local 526 WebRtcRTPHeader rtp_header; local 543 WebRtcRTPHeader rtp_header; local 567 WebRtcRTPHeader rtp_header; local [all...] |
H A D | rtp_receiver_audio.cc | 183 int32_t RTPReceiverAudio::ParseRtpPacket(WebRtcRTPHeader* rtp_header, argument 191 "seqnum", rtp_header->header.sequenceNumber, 192 "timestamp", rtp_header->header.timestamp); 193 rtp_header->type.Audio.numEnergy = rtp_header->header.numCSRCs; 194 num_energy_ = rtp_header->type.Audio.numEnergy; 195 if (rtp_header->type.Audio.numEnergy > 0 && 196 rtp_header->type.Audio.numEnergy <= kRtpCsrcSize) { 198 rtp_header->type.Audio.arrOfEnergy, 199 rtp_header 288 ParseAudioCodecSpecific( WebRtcRTPHeader* rtp_header, const uint8_t* payload_data, uint16_t payload_length, const AudioPayload& audio_specific, bool is_red) argument [all...] |
H A D | rtp_sender_unittest.cc | 47 const uint8_t* GetPayloadData(const RTPHeader& rtp_header, argument 49 return packet + rtp_header.headerLength; 52 uint16_t GetPayloadDataLength(const RTPHeader& rtp_header, argument 54 uint16_t length = packet_length - rtp_header.headerLength - 55 rtp_header.paddingLength; 110 void VerifyRTPHeaderCommon(const RTPHeader& rtp_header) { argument 111 EXPECT_EQ(kMarkerBit, rtp_header.markerBit); 112 EXPECT_EQ(payload_, rtp_header.payloadType); 113 EXPECT_EQ(kSeqNum, rtp_header.sequenceNumber); 114 EXPECT_EQ(kTimestamp, rtp_header 209 webrtc::RTPHeader rtp_header; local 240 webrtc::RTPHeader rtp_header; local 281 webrtc::RTPHeader rtp_header; local 311 webrtc::RTPHeader rtp_header; local 349 webrtc::RTPHeader rtp_header; local 399 webrtc::RTPHeader rtp_header; local 477 webrtc::RTPHeader rtp_header; local 539 webrtc::RTPHeader rtp_header; local 582 webrtc::RTPHeader rtp_header; local 752 webrtc::RTPHeader rtp_header; local 1026 webrtc::RTPHeader rtp_header; local 1055 webrtc::RTPHeader rtp_header; local 1113 webrtc::RTPHeader rtp_header; local [all...] |
H A D | rtp_receiver_impl.cc | 164 const RTPHeader& rtp_header, 173 CheckSSRCChanged(rtp_header); 179 if (CheckPayloadChanged(rtp_header, 198 webrtc_rtp_header.header = rtp_header; 201 uint16_t payload_data_length = payload_length - rtp_header.paddingLength; 208 last_received_sequence_number_ + 1 == rtp_header.sequenceNumber && 209 last_received_timestamp_ != rtp_header.timestamp; 230 if (last_received_timestamp_ != rtp_header.timestamp) { 231 last_received_timestamp_ = rtp_header.timestamp; 234 last_received_sequence_number_ = rtp_header 163 IncomingRtpPacket( const RTPHeader& rtp_header, const uint8_t* payload, int payload_length, PayloadUnion payload_specific, bool in_order) argument 265 CheckSSRCChanged(const RTPHeader& rtp_header) argument 336 CheckPayloadChanged( const RTPHeader& rtp_header, const int8_t first_payload_byte, bool& is_red, PayloadUnion* specific_payload, bool* should_reset_statistics) argument 429 CheckCSRC(const WebRtcRTPHeader& rtp_header) argument [all...] |
H A D | rtp_receiver_video.h | 29 virtual int32_t ParseRtpPacket(WebRtcRTPHeader* rtp_header, 61 int32_t BuildRTPheader(const WebRtcRTPHeader* rtp_header,
|
H A D | rtp_format_h264.cc | 40 void ParseSingleNalu(WebRtcRTPHeader* rtp_header, argument 43 rtp_header->type.Video.codec = kRtpVideoH264; 44 rtp_header->type.Video.isFirstPacket = true; 45 RTPVideoHeaderH264* h264_header = &rtp_header->type.Video.codecHeader.H264; 59 rtp_header->frameType = kVideoFrameKey; 62 rtp_header->frameType = kVideoFrameDelta; 67 void ParseFuaNalu(WebRtcRTPHeader* rtp_header, argument 85 rtp_header->frameType = kVideoFrameKey; 87 rtp_header->frameType = kVideoFrameDelta; 89 rtp_header 297 Parse(WebRtcRTPHeader* rtp_header, const uint8_t* payload_data, size_t payload_data_length) argument [all...] |
H A D | rtp_format_video_generic.cc | 93 bool RtpDepacketizerGeneric::Parse(WebRtcRTPHeader* rtp_header, argument 99 rtp_header->frameType = 103 rtp_header->type.Video.isFirstPacket = 107 payload_data, payload_data_length, rtp_header) != 0) {
|
H A D | rtp_receiver_impl.h | 47 const RTPHeader& rtp_header, 73 void CheckSSRCChanged(const RTPHeader& rtp_header); 74 void CheckCSRC(const WebRtcRTPHeader& rtp_header); 75 int32_t CheckPayloadChanged(const RTPHeader& rtp_header,
|
H A D | fec_receiver_impl.h | 31 virtual int32_t AddReceivedRedPacket(const RTPHeader& rtp_header,
|
H A D | rtp_format_vp8.cc | 131 int ParseVP8FrameSize(WebRtcRTPHeader* rtp_header, argument 134 assert(rtp_header != NULL); 135 if (rtp_header->frameType != kVideoFrameKey) { 144 rtp_header->type.Video.width = ((data[7] << 8) + data[6]) & 0x3FFF; 145 rtp_header->type.Video.height = ((data[9] << 8) + data[8]) & 0x3FFF; 173 bool ParseVP8(WebRtcRTPHeader* rtp_header, argument 177 assert(rtp_header != NULL); 183 rtp_header->type.Video.isFirstPacket = 186 rtp_header->type.Video.codecHeader.VP8.nonReference = 188 rtp_header [all...] |
H A D | rtp_sender.cc | 478 RTPHeader rtp_header; local 479 rtp_parser.Parse(rtp_header); 480 bytes_left -= length - rtp_header.headerLength; 578 RTPHeader rtp_header; local 579 rtp_parser.Parse(rtp_header); 583 padding_packet, length, rtp_header, now_ms - capture_time_ms); 586 UpdateAbsoluteSendTime(padding_packet, length, rtp_header, now_ms); 590 UpdateRtpStats(padding_packet, length, rtp_header, over_rtx, false); 820 RTPHeader rtp_header; local 821 rtp_parser.Parse(rtp_header); 916 RTPHeader rtp_header; local 1267 UpdateTransmissionTimeOffset( uint8_t *rtp_packet, const uint16_t rtp_packet_length, const RTPHeader &rtp_header, const int64_t time_diff_ms) const argument 1313 UpdateAudioLevel(uint8_t *rtp_packet, const uint16_t rtp_packet_length, const RTPHeader &rtp_header, const bool is_voiced, const uint8_t dBov) const argument 1356 UpdateAbsoluteSendTime( uint8_t *rtp_packet, const uint16_t rtp_packet_length, const RTPHeader &rtp_header, const int64_t now_ms) const argument 1632 RTPHeader rtp_header; local [all...] |
/external/chromium_org/media/cast/net/rtp/ |
H A D | rtp_packetizer_unittest.cc | 42 void VerifyRtpHeader(const RtpCastTestHeader& rtp_header) { argument 43 VerifyCommonRtpHeader(rtp_header); 44 VerifyCastRtpHeader(rtp_header); 47 void VerifyCommonRtpHeader(const RtpCastTestHeader& rtp_header) { argument 48 EXPECT_EQ(kPayload, rtp_header.payload_type); 49 EXPECT_EQ(sequence_number_, rtp_header.sequence_number); 50 EXPECT_EQ(expectd_rtp_timestamp_, rtp_header.rtp_timestamp); 51 EXPECT_EQ(config_.ssrc, rtp_header.ssrc); 52 EXPECT_EQ(0, rtp_header.num_csrcs); 55 void VerifyCastRtpHeader(const RtpCastTestHeader& rtp_header) { argument 67 RtpCastTestHeader rtp_header; variable 69 VerifyRtpHeader(rtp_header); variable [all...] |
H A D | frame_buffer.cc | 27 const RtpCastHeader& rtp_header) { 30 frame_id_ = rtp_header.frame_id; 31 max_packet_id_ = rtp_header.max_packet_id; 32 is_key_frame_ = rtp_header.is_key_frame; 33 new_playout_delay_ms_ = rtp_header.new_playout_delay_ms; 35 DCHECK_EQ(rtp_header.frame_id, rtp_header.reference_frame_id); 36 last_referenced_frame_id_ = rtp_header.reference_frame_id; 37 rtp_timestamp_ = rtp_header.rtp_timestamp; 40 if (rtp_header 25 InsertPacket(const uint8* payload_data, size_t payload_size, const RtpCastHeader& rtp_header) argument [all...] |
H A D | rtp_parser.h | 25 // pointed to by |rtp_header| and sets the |payload_data| pointer and 32 RtpCastHeader* rtp_header,
|
H A D | framer.cc | 37 const RtpCastHeader& rtp_header, 40 uint32 frame_id = rtp_header.frame_id; 42 if (rtp_header.is_key_frame && waiting_for_key_) { 48 << " packet:" << static_cast<int>(rtp_header.packet_id) 49 << " max packet:" << static_cast<int>(rtp_header.max_packet_id); 72 if (!it->second->InsertPacket(payload_data, payload_size, rtp_header)) { 74 << static_cast<int>(rtp_header.frame_id) << ", packet " 75 << rtp_header.packet_id; 35 InsertPacket(const uint8* payload_data, size_t payload_size, const RtpCastHeader& rtp_header, bool* duplicate) argument
|
/external/chromium_org/third_party/webrtc/modules/audio_coding/neteq/ |
H A D | rtcp.cc | 33 void Rtcp::Update(const RTPHeader& rtp_header, uint32_t receive_timestamp) { argument 36 int16_t sn_diff = rtp_header.sequenceNumber - max_seq_no_; 38 if (rtp_header.sequenceNumber < max_seq_no_) { 42 max_seq_no_ = rtp_header.sequenceNumber; 48 int32_t ts_diff = receive_timestamp - (rtp_header.timestamp - transit_); 54 transit_ = rtp_header.timestamp - receive_timestamp;
|
H A D | neteq_impl_unittest.cc | 261 WebRtcRTPHeader rtp_header; local 262 rtp_header.header.payloadType = kPayloadType; 263 rtp_header.header.sequenceNumber = kFirstSequenceNumber; 264 rtp_header.header.timestamp = kFirstTimestamp; 265 rtp_header.header.ssrc = kSsrc; 320 .WillOnce(Return(&rtp_header.header)); 354 neteq_->InsertPacket(rtp_header, payload, kPayloadLength, kFirstReceiveTime); 357 rtp_header.header.timestamp += 160; 358 rtp_header.header.sequenceNumber += 1; 359 neteq_->InsertPacket(rtp_header, payloa 372 WebRtcRTPHeader rtp_header; local 414 WebRtcRTPHeader rtp_header; local [all...] |
H A D | rtcp.h | 35 void Update(const RTPHeader& rtp_header, uint32_t receive_timestamp);
|
/external/chromium_org/third_party/webrtc/modules/rtp_rtcp/interface/ |
H A D | fec_receiver.h | 25 virtual int32_t AddReceivedRedPacket(const RTPHeader& rtp_header,
|
/external/chromium_org/third_party/webrtc/modules/video_coding/main/test/ |
H A D | receiver_tests.h | 33 const webrtc::WebRtcRTPHeader* rtp_header) OVERRIDE { 34 return vcm_->IncomingPacket(payload_data, payload_size, *rtp_header);
|
/external/chromium_org/media/cast/receiver/ |
H A D | frame_receiver.cc | 79 RtpCastHeader rtp_header; local 84 &rtp_header, 90 ProcessParsedPacket(rtp_header, payload_data, payload_size); 91 stats_.UpdateStatistics(rtp_header); 112 void FrameReceiver::ProcessParsedPacket(const RtpCastHeader& rtp_header, argument 119 frame_id_to_rtp_timestamp_[rtp_header.frame_id & 0xff] = 120 rtp_header.rtp_timestamp; 122 now, PACKET_RECEIVED, event_media_type_, rtp_header.rtp_timestamp, 123 rtp_header.frame_id, rtp_header [all...] |